The call voice quality is one of the critical indexes for measuring the telecom network, and also a core competence of the telecom equipment manufacturers. In order to ensure the voice quality during the call, a key point is to effectively process the echoes and other interference sources generated during the call, so as to improve the subjective feelings of both parties concerned in the call.
The echoes generated during the call may be classified into electric echo and acoustic echo according to their sources. In which, the mechanism for producing the electric echo is illustrated in FIG. 1. Since the working state of the 2/4 wire hybrid in the telecom network is closely related to the characteristics of the external line impedance, the transmitting end and the receiving end cannot be completely isolated from each other when the hybrid does not match the external line impedance. Thus, as illustrated in FIG. 1, different phones and different lengths of subscriber lines cause a signal leakage at the hybrid, and the voice of the subscriber can be heard by the counterpart subscriber. The mechanism for producing the acoustic echo is illustrated in FIG. 2, mainly owing to the coupling between the loudspeaker and the microphone. Referring to FIG. 2, the far-end acoustic signal played by the near-end loudspeaker always has a part transmitted back to the far-end through the microphone, and the far-end subscriber hears his own voice again.
The conventional echo control algorithm is implemented based on the Pulse Code Modulation (PCM) in the linear domain. The so called linear domain is a mode based on direct sample point value, and when the communication link has a codec compression part, corresponding decoder is required to completely decode the compressed code stream and recover to the sample point value. It is adaptive to the scenario where the passed network elements require a decoding or the input is PCM in the linear domain. However, with the occurrences of the Tandem Free Operation (TFO) and the Transcoder Free Operation (TrFO) in recent years, it is not required to perform multiple times of transforms of speech codec between the transmitting and receiving ends, thus the conventional echo control algorithm is no longer applicable, and characteristic parameters in the parameter domain shall be obtained through decoding in the parameter domain to perform an echo control in the parameter domain. The so called decoding in the parameter domain means only partially decoding the compressed code stream to extract the characteristic parameters in the parameter domain of each frame, including fixed codebook gain, adaptive codebook gain, line spectrum frequency, etc. The echo control is implemented by modifying the characteristic parameters in the parameter domain in the compressed code stream, without recovering the compressed code stream to the sample point value through the decoder.
When an echo control is performed in the parameter domain, the input and output are all compressed code signals, thus for some encoding types such as Adaptive Multi Rate Codec (AMR) and Enhanced Full Rate Codec (EFR), the frames are associated with each other, and it shall not simply and independently perform the echo control for a certain frame. Particularly, during a handover between echo and non-echo, some special processing is required to achieve a more natural transition between echo and non-echo.
The prior art provides a method for echo control applicable to TFO/TrFO scenarios. The method performs a transition between echo and non-echo by re-quantizing and outputting the fixed codebook gain and the adaptive codebook gain in the near-end input signal. FIG. 3 illustrates a structure diagram of an apparatus provided by the prior art that implements an echo control in the parameter domain through the above method.
However, during the transition between echo and non-echo in the prior art, when related processing of the code type signals such as AMR and EFR are to be made, the quantization of a parameter is associated with the prediction error of the parameter in the previous frame because the frames of those signals are associated with each other, while the prior art does not re-quantize the related linear prediction coefficients. Thus when the signals using the above coding mode are processed in the prior art, there is a risk that the finally decoded linear prediction parameters may be abnormal, resulting in signal mutations, then the handover between echo and non-echo cannot be smoothly transited.